#include "avformat.h"
#include <stdarg.h>
-#include <netinet/in.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/wait.h>
+#include <netinet/in.h>
#include <arpa/inet.h>
#include <netdb.h>
-#include <ctype.h>
#include <signal.h>
+#ifdef CONFIG_HAVE_DLFCN
#include <dlfcn.h>
+#endif
#include "ffserver.h"
RTSPSTATE_WAIT_REQUEST,
RTSPSTATE_SEND_REPLY,
+ RTSPSTATE_SEND_PACKET,
};
const char *http_state[] = {
"RTSP_WAIT_REQUEST",
"RTSP_SEND_REPLY",
+ "RTSP_SEND_PACKET",
};
#define IOBUFFER_INIT_SIZE 8192
#define SYNC_TIMEOUT (10 * 1000)
typedef struct {
- INT64 count1, count2;
+ int64_t count1, count2;
long time1, time2;
} DataRateData;
struct sockaddr_in from_addr; /* origin */
struct pollfd *poll_entry; /* used when polling */
long timeout;
- UINT8 *buffer_ptr, *buffer_end;
+ uint8_t *buffer_ptr, *buffer_end;
int http_error;
struct HTTPContext *next;
int got_key_frame; /* stream 0 => 1, stream 1 => 2, stream 2=> 4 */
- INT64 data_count;
+ int64_t data_count;
/* feed input */
int feed_fd;
/* input format handling */
AVFormatContext *fmt_in;
long start_time; /* In milliseconds - this wraps fairly often */
- INT64 first_pts; /* initial pts value */
+ int64_t first_pts; /* initial pts value */
+ int64_t cur_pts; /* current pts value */
int pts_stream_index; /* stream we choose as clock reference */
/* output format handling */
struct FFStream *stream;
AVFormatContext fmt_ctx; /* instance of FFStream for one user */
int last_packet_sent; /* true if last data packet was sent */
int suppress_log;
- int bandwidth;
DataRateData datarate;
int wmp_client_id;
char protocol[16];
char method[16];
char url[128];
int buffer_size;
- UINT8 *buffer;
+ uint8_t *buffer;
int is_packetized; /* if true, the stream is packetized */
int packet_stream_index; /* current stream for output in state machine */
/* RTSP state specific */
- UINT8 *pb_buffer; /* XXX: use that in all the code */
+ uint8_t *pb_buffer; /* XXX: use that in all the code */
ByteIOContext *pb;
int seq; /* RTSP sequence number */
enum RTSPProtocol rtp_protocol;
char session_id[32]; /* session id */
AVFormatContext *rtp_ctx[MAX_STREAMS];
- URLContext *rtp_handles[MAX_STREAMS];
/* RTP short term bandwidth limitation */
int packet_byte_count;
int packet_start_time_us; /* used for short durations (a few
seconds max) */
+ /* RTP/UDP specific */
+ URLContext *rtp_handles[MAX_STREAMS];
+
+ /* RTP/TCP specific */
+ struct HTTPContext *rtsp_c;
+ uint8_t *packet_buffer, *packet_buffer_ptr, *packet_buffer_end;
} HTTPContext;
+static AVFrame dummy_frame;
+
/* each generated stream is described here */
enum StreamType {
STREAM_TYPE_LIVE,
typedef struct IPAddressACL {
struct IPAddressACL *next;
enum IPAddressAction action;
+ /* These are in host order */
struct in_addr first;
struct in_addr last;
} IPAddressACL;
time_t pid_start; /* Of ffmpeg process */
char **child_argv;
struct FFStream *next;
+ int bandwidth; /* bandwidth, in kbits/s */
/* RTSP options */
char *rtsp_option;
+ /* multicast specific */
+ int is_multicast;
+ struct in_addr multicast_ip;
+ int multicast_port; /* first port used for multicast */
+ int multicast_ttl;
+ int loop; /* if true, send the stream in loops (only meaningful if file) */
+
/* feed specific */
int feed_opened; /* true if someone is writing to the feed */
int is_feed; /* true if it is a feed */
+ int readonly; /* True if writing is prohibited to the file */
int conns_served;
- INT64 bytes_served;
- INT64 feed_max_size; /* maximum storage size */
- INT64 feed_write_index; /* current write position in feed (it wraps round) */
- INT64 feed_size; /* current size of feed */
+ int64_t bytes_served;
+ int64_t feed_max_size; /* maximum storage size */
+ int64_t feed_write_index; /* current write position in feed (it wraps round) */
+ int64_t feed_size; /* current size of feed */
struct FFStream *next_feed;
} FFStream;
/* RTSP handling */
static int rtsp_parse_request(HTTPContext *c);
static void rtsp_cmd_describe(HTTPContext *c, const char *url);
+static void rtsp_cmd_options(HTTPContext *c, const char *url);
static void rtsp_cmd_setup(HTTPContext *c, const char *url, RTSPHeader *h);
static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h);
static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPHeader *h);
static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h);
+/* SDP handling */
+static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
+ struct in_addr my_ip);
+
/* RTP handling */
-static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
- FFStream *stream, const char *session_id);
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol);
static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr);
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c);
static const char *my_program_name;
static const char *my_program_dir;
int nb_max_connections;
int nb_connections;
-int nb_max_bandwidth;
-int nb_bandwidth;
+int max_bandwidth;
+int current_bandwidth;
static long cur_time; // Making this global saves on passing it around everywhere
static FILE *logfile = NULL;
-static void http_log(char *fmt, ...)
+static void __attribute__ ((format (printf, 1, 2))) http_log(const char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
va_end(ap);
}
-static void log_connection(HTTPContext *c)
+static char *ctime1(char *buf2)
{
- char buf1[32], buf2[32], *p;
time_t ti;
+ char *p;
- if (c->suppress_log)
- return;
-
- /* XXX: reentrant function ? */
- p = inet_ntoa(c->from_addr.sin_addr);
- strcpy(buf1, p);
ti = time(NULL);
p = ctime(&ti);
strcpy(buf2, p);
p = buf2 + strlen(p) - 1;
if (*p == '\n')
*p = '\0';
+ return buf2;
+}
+
+static void log_connection(HTTPContext *c)
+{
+ char buf2[32];
+
+ if (c->suppress_log)
+ return;
+
http_log("%s - - [%s] \"%s %s %s\" %d %lld\n",
- buf1, buf2, c->method, c->url, c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
+ inet_ntoa(c->from_addr.sin_addr),
+ ctime1(buf2), c->method, c->url,
+ c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
}
-static void update_datarate(DataRateData *drd, INT64 count)
+static void update_datarate(DataRateData *drd, int64_t count)
{
if (!drd->time1 && !drd->count1) {
drd->time1 = drd->time2 = cur_time;
}
/* In bytes per second */
-static int compute_datarate(DataRateData *drd, INT64 count)
+static int compute_datarate(DataRateData *drd, int64_t count)
{
if (cur_time == drd->time1)
return 0;
-
+
return ((count - drd->count1) * 1000) / (cur_time - drd->time1);
}
-static int get_longterm_datarate(DataRateData *drd, INT64 count)
+static int get_longterm_datarate(DataRateData *drd, int64_t count)
{
/* You get the first 3 seconds flat out */
if (cur_time - drd->time1 < 3000)
return 0;
-
return compute_datarate(drd, count);
}
/* This is needed to make relative pathnames work */
chdir(my_program_dir);
+ signal(SIGPIPE, SIG_DFL);
+
execvp(pathname, feed->child_argv);
_exit(1);
setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp));
if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) {
- perror ("bind");
+ char bindmsg[32];
+ snprintf(bindmsg, sizeof(bindmsg), "bind(port %d)", ntohs(my_addr->sin_port));
+ perror (bindmsg);
close(server_fd);
return -1;
}
return server_fd;
}
+/* start all multicast streams */
+static void start_multicast(void)
+{
+ FFStream *stream;
+ char session_id[32];
+ HTTPContext *rtp_c;
+ struct sockaddr_in dest_addr;
+ int default_port, stream_index;
+
+ default_port = 6000;
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ if (stream->is_multicast) {
+ /* open the RTP connection */
+ snprintf(session_id, sizeof(session_id),
+ "%08x%08x", (int)random(), (int)random());
+
+ /* choose a port if none given */
+ if (stream->multicast_port == 0) {
+ stream->multicast_port = default_port;
+ default_port += 100;
+ }
+
+ dest_addr.sin_family = AF_INET;
+ dest_addr.sin_addr = stream->multicast_ip;
+ dest_addr.sin_port = htons(stream->multicast_port);
+
+ rtp_c = rtp_new_connection(&dest_addr, stream, session_id,
+ RTSP_PROTOCOL_RTP_UDP_MULTICAST);
+ if (!rtp_c) {
+ continue;
+ }
+ if (open_input_stream(rtp_c, "") < 0) {
+ fprintf(stderr, "Could not open input stream for stream '%s'\n",
+ stream->filename);
+ continue;
+ }
+
+ /* open each RTP stream */
+ for(stream_index = 0; stream_index < stream->nb_streams;
+ stream_index++) {
+ dest_addr.sin_port = htons(stream->multicast_port +
+ 2 * stream_index);
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, NULL) < 0) {
+ fprintf(stderr, "Could not open output stream '%s/streamid=%d'\n",
+ stream->filename, stream_index);
+ exit(1);
+ }
+ }
+
+ /* change state to send data */
+ rtp_c->state = HTTPSTATE_SEND_DATA;
+ }
+ }
+}
/* main loop of the http server */
static int http_server(void)
first_http_ctx = NULL;
nb_connections = 0;
- first_http_ctx = NULL;
+
+ start_multicast();
+
for(;;) {
poll_entry = poll_table;
poll_entry->fd = server_fd;
switch(c->state) {
case HTTPSTATE_SEND_HEADER:
case RTSPSTATE_SEND_REPLY:
+ case RTSPSTATE_SEND_PACKET:
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLOUT;
if (!c)
goto fail;
- c->next = first_http_ctx;
- first_http_ctx = c;
c->fd = fd;
c->poll_entry = NULL;
c->from_addr = from_addr;
c->buffer = av_malloc(c->buffer_size);
if (!c->buffer)
goto fail;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
nb_connections++;
start_wait_request(c, is_rtsp);
}
}
+ /* remove references, if any (XXX: do it faster) */
+ for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
+ if (c1->rtsp_c == c)
+ c1->rtsp_c = NULL;
+ }
+
/* remove connection associated resources */
if (c->fd >= 0)
close(c->fd);
url_close(h);
}
}
+
+ ctx = &c->fmt_ctx;
+
+ if (!c->last_packet_sent) {
+ if (ctx->oformat) {
+ /* prepare header */
+ if (url_open_dyn_buf(&ctx->pb) >= 0) {
+ av_write_trailer(ctx);
+ url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ }
+ }
+ }
+
+ for(i=0; i<ctx->nb_streams; i++)
+ av_free(ctx->streams[i]) ;
- nb_bandwidth -= c->bandwidth;
+ if (c->stream)
+ current_bandwidth -= c->stream->bandwidth;
av_freep(&c->pb_buffer);
+ av_freep(&c->packet_buffer);
av_free(c->buffer);
av_free(c);
nb_connections--;
if (!(c->poll_entry->revents & POLLIN))
return 0;
/* read the data */
- len = read(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ read_loop:
+ len = read(c->fd, c->buffer_ptr, 1);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR)
return -1;
} else if (len == 0) {
return -1;
} else {
- /* search for end of request. XXX: not fully correct since garbage could come after the end */
- UINT8 *ptr;
+ /* search for end of request. */
+ uint8_t *ptr;
c->buffer_ptr += len;
ptr = c->buffer_ptr;
if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
return -1;
- }
+ } else goto read_loop;
}
break;
}
}
break;
+ case RTSPSTATE_SEND_PACKET:
+ if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ /* no need to write if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ len = write(c->fd, c->packet_buffer_ptr,
+ c->packet_buffer_end - c->packet_buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ } else {
+ c->packet_buffer_ptr += len;
+ if (c->packet_buffer_ptr >= c->packet_buffer_end) {
+ /* all the buffer was sent : wait for a new request */
+ av_freep(&c->packet_buffer);
+ c->state = RTSPSTATE_WAIT_REQUEST;
+ }
+ }
+ break;
case HTTPSTATE_READY:
/* nothing to do */
break;
FFStream *req = c->stream;
int action_required = 0;
+ /* Not much we can do for a feed */
+ if (!req->feed)
+ return 0;
+
for (i = 0; i < req->nb_streams; i++) {
AVCodecContext *codec = &req->streams[i]->codec;
enum IPAddressAction last_action = IP_DENY;
IPAddressACL *acl;
struct in_addr *src = &c->from_addr.sin_addr;
+ unsigned long src_addr = ntohl(src->s_addr);
for (acl = stream->acl; acl; acl = acl->next) {
- if (src->s_addr >= acl->first.s_addr && src->s_addr <= acl->last.s_addr) {
+ if (src_addr >= acl->first.s_addr && src_addr <= acl->last.s_addr) {
return (acl->action == IP_ALLOW) ? 1 : 0;
}
last_action = acl->action;
return (last_action == IP_DENY) ? 1 : 0;
}
+/* compute the real filename of a file by matching it without its
+ extensions to all the stream filenames */
+static void compute_real_filename(char *filename, int max_size)
+{
+ char file1[1024];
+ char file2[1024];
+ char *p;
+ FFStream *stream;
+
+ /* compute filename by matching without the file extensions */
+ pstrcpy(file1, sizeof(file1), filename);
+ p = strrchr(file1, '.');
+ if (p)
+ *p = '\0';
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ pstrcpy(file2, sizeof(file2), stream->filename);
+ p = strrchr(file2, '.');
+ if (p)
+ *p = '\0';
+ if (!strcmp(file1, file2)) {
+ pstrcpy(filename, max_size, stream->filename);
+ break;
+ }
+ }
+}
+
+enum RedirType {
+ REDIR_NONE,
+ REDIR_ASX,
+ REDIR_RAM,
+ REDIR_ASF,
+ REDIR_RTSP,
+ REDIR_SDP,
+};
+
/* parse http request and prepare header */
static int http_parse_request(HTTPContext *c)
{
char *p;
int post;
- int doing_asx;
- int doing_asf_redirector;
- int doing_ram;
- int doing_rtsp_redirector;
+ enum RedirType redir_type;
char cmd[32];
char info[1024], *filename;
char url[1024], *q;
p++;
}
- if (strlen(filename) > 4 && strcmp(".asx", filename + strlen(filename) - 4) == 0) {
- doing_asx = 1;
+ redir_type = REDIR_NONE;
+ if (match_ext(filename, "asx")) {
+ redir_type = REDIR_ASX;
filename[strlen(filename)-1] = 'f';
- } else {
- doing_asx = 0;
- }
-
- if (strlen(filename) > 4 && strcmp(".asf", filename + strlen(filename) - 4) == 0 &&
+ } else if (match_ext(filename, "asf") &&
(!useragent || strncasecmp(useragent, "NSPlayer", 8) != 0)) {
/* if this isn't WMP or lookalike, return the redirector file */
- doing_asf_redirector = 1;
- } else {
- doing_asf_redirector = 0;
- }
-
- if (strlen(filename) > 4 &&
- (strcmp(".rpm", filename + strlen(filename) - 4) == 0 ||
- strcmp(".ram", filename + strlen(filename) - 4) == 0)) {
- doing_ram = 1;
+ redir_type = REDIR_ASF;
+ } else if (match_ext(filename, "rpm,ram")) {
+ redir_type = REDIR_RAM;
strcpy(filename + strlen(filename)-2, "m");
- } else {
- doing_ram = 0;
- }
-
- if (strlen(filename) > 5 &&
- strcmp(".rtsp", filename + strlen(filename) - 5) == 0) {
- char file1[1024];
- char file2[1024];
- char *p;
-
- doing_rtsp_redirector = 1;
- /* compute filename by matching without the file extensions */
- pstrcpy(file1, sizeof(file1), filename);
- p = strrchr(file1, '.');
- if (p)
- *p = '\0';
- for(stream = first_stream; stream != NULL; stream = stream->next) {
- pstrcpy(file2, sizeof(file2), stream->filename);
- p = strrchr(file2, '.');
- if (p)
- *p = '\0';
- if (!strcmp(file1, file2)) {
- pstrcpy(url, sizeof(url), stream->filename);
- filename = url;
- break;
- }
- }
- } else {
- doing_rtsp_redirector = 0;
+ } else if (match_ext(filename, "rtsp")) {
+ redir_type = REDIR_RTSP;
+ compute_real_filename(filename, sizeof(url) - 1);
+ } else if (match_ext(filename, "sdp")) {
+ redir_type = REDIR_SDP;
+ compute_real_filename(filename, sizeof(url) - 1);
}
-
+
stream = first_stream;
while (stream != NULL) {
if (!strcmp(stream->filename, filename) && validate_acl(stream, c))
}
if (post == 0 && stream->stream_type == STREAM_TYPE_LIVE) {
- /* See if we meet the bandwidth requirements */
- for(i=0;i<stream->nb_streams;i++) {
- AVStream *st = stream->streams[i];
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- c->bandwidth += st->codec.bit_rate;
- break;
- case CODEC_TYPE_VIDEO:
- c->bandwidth += st->codec.bit_rate;
- break;
- default:
- av_abort();
- }
- }
+ current_bandwidth += stream->bandwidth;
}
-
- c->bandwidth /= 1000;
- nb_bandwidth += c->bandwidth;
-
- if (post == 0 && nb_max_bandwidth < nb_bandwidth) {
+
+ if (post == 0 && max_bandwidth < current_bandwidth) {
c->http_error = 200;
q = c->buffer;
q += sprintf(q, "HTTP/1.0 200 Server too busy\r\n");
q += sprintf(q, "<html><head><title>Too busy</title></head><body>\r\n");
q += sprintf(q, "The server is too busy to serve your request at this time.<p>\r\n");
q += sprintf(q, "The bandwidth being served (including your stream) is %dkbit/sec, and this exceeds the limit of %dkbit/sec\r\n",
- nb_bandwidth, nb_max_bandwidth);
+ current_bandwidth, max_bandwidth);
q += sprintf(q, "</body></html>\r\n");
/* prepare output buffer */
return 0;
}
- if (doing_asx || doing_ram || doing_asf_redirector ||
- doing_rtsp_redirector) {
+ if (redir_type != REDIR_NONE) {
char *hostinfo = 0;
for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
c->http_error = 200;
q = c->buffer;
- if (doing_asx) {
+ switch(redir_type) {
+ case REDIR_ASX:
q += sprintf(q, "HTTP/1.0 200 ASX Follows\r\n");
q += sprintf(q, "Content-type: video/x-ms-asf\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "<ENTRY><REF HREF=\"http://%s/%s%s\"/></ENTRY>\r\n",
hostbuf, filename, info);
q += sprintf(q, "</ASX>\r\n");
- } else if (doing_ram) {
+ break;
+ case REDIR_RAM:
q += sprintf(q, "HTTP/1.0 200 RAM Follows\r\n");
q += sprintf(q, "Content-type: audio/x-pn-realaudio\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "# Autogenerated by ffserver\r\n");
q += sprintf(q, "http://%s/%s%s\r\n",
hostbuf, filename, info);
- } else if (doing_asf_redirector) {
+ break;
+ case REDIR_ASF:
q += sprintf(q, "HTTP/1.0 200 ASF Redirect follows\r\n");
q += sprintf(q, "Content-type: video/x-ms-asf\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "[Reference]\r\n");
q += sprintf(q, "Ref1=http://%s/%s%s\r\n",
hostbuf, filename, info);
- } else if (doing_rtsp_redirector) {
- char hostname[256], *p;
- /* extract only hostname */
- pstrcpy(hostname, sizeof(hostname), hostbuf);
- p = strrchr(hostname, ':');
- if (p)
- *p = '\0';
- q += sprintf(q, "HTTP/1.0 200 RTSP Redirect follows\r\n");
- /* XXX: incorrect mime type ? */
- q += sprintf(q, "Content-type: application/x-rtsp\r\n");
- q += sprintf(q, "\r\n");
- q += sprintf(q, "rtsp://%s:%d/%s\r\n",
- hostname, ntohs(my_rtsp_addr.sin_port),
- filename);
- } else {
+ break;
+ case REDIR_RTSP:
+ {
+ char hostname[256], *p;
+ /* extract only hostname */
+ pstrcpy(hostname, sizeof(hostname), hostbuf);
+ p = strrchr(hostname, ':');
+ if (p)
+ *p = '\0';
+ q += sprintf(q, "HTTP/1.0 200 RTSP Redirect follows\r\n");
+ /* XXX: incorrect mime type ? */
+ q += sprintf(q, "Content-type: application/x-rtsp\r\n");
+ q += sprintf(q, "\r\n");
+ q += sprintf(q, "rtsp://%s:%d/%s\r\n",
+ hostname, ntohs(my_rtsp_addr.sin_port),
+ filename);
+ }
+ break;
+ case REDIR_SDP:
+ {
+ uint8_t *sdp_data;
+ int sdp_data_size, len;
+ struct sockaddr_in my_addr;
+
+ q += sprintf(q, "HTTP/1.0 200 OK\r\n");
+ q += sprintf(q, "Content-type: application/sdp\r\n");
+ q += sprintf(q, "\r\n");
+
+ len = sizeof(my_addr);
+ getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
+
+ /* XXX: should use a dynamic buffer */
+ sdp_data_size = prepare_sdp_description(stream,
+ &sdp_data,
+ my_addr.sin_addr);
+ if (sdp_data_size > 0) {
+ memcpy(q, sdp_data, sdp_data_size);
+ q += sdp_data_size;
+ *q = '\0';
+ av_free(sdp_data);
+ }
+ }
+ break;
+ default:
av_abort();
+ break;
}
/* prepare output buffer */
}
sprintf(msg, "POST command not handled");
+ c->stream = 0;
goto send_error;
}
if (http_start_receive_data(c) < 0) {
c->wmp_client_id = random() & 0x7fffffff;
q += sprintf(q, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id);
- mime_type = "application/octet-stream";
}
q += sprintf(q, "Content-Type: %s\r\n", mime_type);
q += sprintf(q, "\r\n");
return 0;
}
-static void fmt_bytecount(ByteIOContext *pb, INT64 count)
+static void fmt_bytecount(ByteIOContext *pb, int64_t count)
{
static const char *suffix = " kMGTP";
const char *s;
} else if (strcmp(eosf - 3, ".rm") == 0) {
strcpy(eosf - 3, ".ram");
} else if (stream->fmt == &rtp_mux) {
- /* generate a sample RTSP director - maybe should
- generate a .sdp file ? */
+ /* generate a sample RTSP director if
+ unicast. Generate an SDP redirector if
+ multicast */
eosf = strrchr(sfilename, '.');
if (!eosf)
eosf = sfilename + strlen(sfilename);
- strcpy(eosf, ".rtsp");
+ if (stream->is_multicast)
+ strcpy(eosf, ".sdp");
+ else
+ strcpy(eosf, ".rtsp");
}
}
{
int audio_bit_rate = 0;
int video_bit_rate = 0;
- char *audio_codec_name = "";
- char *video_codec_name = "";
- char *audio_codec_name_extra = "";
- char *video_codec_name_extra = "";
+ const char *audio_codec_name = "";
+ const char *video_codec_name = "";
+ const char *audio_codec_name_extra = "";
+ const char *video_codec_name_extra = "";
for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
}
url_fprintf(pb, "<TD align=center> %s <TD align=right> %d <TD align=right> %d <TD> %s %s <TD align=right> %d <TD> %s %s",
stream->fmt->name,
- (audio_bit_rate + video_bit_rate) / 1000,
+ stream->bandwidth,
video_bit_rate / 1000, video_codec_name, video_codec_name_extra,
audio_bit_rate / 1000, audio_codec_name, audio_codec_name_extra);
if (stream->feed) {
for (i = 0; i < stream->nb_streams; i++) {
AVStream *st = stream->streams[i];
AVCodec *codec = avcodec_find_encoder(st->codec.codec_id);
- char *type = "unknown";
+ const char *type = "unknown";
char parameters[64];
parameters[0] = 0;
case CODEC_TYPE_VIDEO:
type = "video";
sprintf(parameters, "%dx%d, q=%d-%d, fps=%d", st->codec.width, st->codec.height,
- st->codec.qmin, st->codec.qmax, st->codec.frame_rate / FRAME_RATE_BASE);
+ st->codec.qmin, st->codec.qmax, st->codec.frame_rate / st->codec.frame_rate_base);
break;
default:
av_abort();
nb_connections, nb_max_connections);
url_fprintf(pb, "Bandwidth in use: %dk / %dk<BR>\n",
- nb_bandwidth, nb_max_bandwidth);
+ current_bandwidth, max_bandwidth);
url_fprintf(pb, "<TABLE>\n");
url_fprintf(pb, "<TR><th>#<th>File<th>IP<th>Proto<th>State<th>Target bits/sec<th>Actual bits/sec<th>Bytes transferred\n");
char input_filename[1024];
AVFormatContext *s;
int buf_size, i;
- INT64 stream_pos;
+ int64_t stream_pos;
/* find file name */
if (c->stream->feed) {
stream_pos = parse_date(buf, 0);
} else if (find_info_tag(buf, sizeof(buf), "buffer", info)) {
int prebuffer = strtol(buf, 0, 10);
- stream_pos = av_gettime() - prebuffer * (INT64)1000000;
+ stream_pos = av_gettime() - prebuffer * (int64_t)1000000;
} else {
- stream_pos = av_gettime() - c->stream->prebuffer * (INT64)1000;
+ stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000;
}
} else {
strcpy(input_filename, c->stream->feed_filename);
if (st->pts.den == 0) {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
- st->pts_incr = (INT64)s->pts_den;
+ st->pts_incr = (int64_t)s->pts_den;
av_frac_init(&st->pts, st->pts.val, 0,
- (INT64)s->pts_num * st->codec.sample_rate);
+ (int64_t)s->pts_num * st->codec.sample_rate);
break;
case CODEC_TYPE_VIDEO:
- st->pts_incr = (INT64)s->pts_den * FRAME_RATE_BASE;
+ st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
av_frac_init(&st->pts, st->pts.val, 0,
- (INT64)s->pts_num * st->codec.frame_rate);
+ (int64_t)s->pts_num * st->codec.frame_rate);
break;
default:
av_abort();
/* we use the codec indication because it is
more accurate than the demux flags */
pkt->flags = 0;
- if (st->codec.key_frame)
+ if (st->codec.coded_frame->key_frame)
pkt->flags |= PKT_FLAG_KEY;
return 0;
}
if (st->pts.den == 0) {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
- st->pts_incr = (INT64)s->pts_den * st->codec.frame_size;
+ st->pts_incr = (int64_t)s->pts_den * st->codec.frame_size;
av_frac_init(&st->pts, st->pts.val, 0,
- (INT64)s->pts_num * st->codec.sample_rate);
+ (int64_t)s->pts_num * st->codec.sample_rate);
break;
case CODEC_TYPE_VIDEO:
- st->pts_incr = (INT64)s->pts_den * FRAME_RATE_BASE;
+ st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
av_frac_init(&st->pts, st->pts.val, 0,
- (INT64)s->pts_num * st->codec.frame_rate);
+ (int64_t)s->pts_num * st->codec.frame_rate);
break;
default:
av_abort();
static int compute_send_delay(HTTPContext *c)
{
- INT64 cur_pts, delta_pts, next_pts;
+ int64_t cur_pts, delta_pts, next_pts;
int delay1;
/* compute current pts value from system time */
- cur_pts = ((INT64)(cur_time - c->start_time) * c->fmt_in->pts_den) /
+ cur_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
(c->fmt_in->pts_num * 1000LL);
/* compute the delta from the stream we choose as
main clock (we do that to avoid using explicit
#else
/* just fall backs */
-int av_read_frame(AVFormatContext *s, AVPacket *pkt)
+static int av_read_frame(AVFormatContext *s, AVPacket *pkt)
{
return av_read_packet(s, pkt);
}
static int compute_send_delay(HTTPContext *c)
{
int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
+ int64_t delta_pts;
+ int64_t time_pts;
+ int m_delay;
- if (datarate > c->bandwidth * 2000) {
+ if (datarate > c->stream->bandwidth * 2000) {
return 1000;
}
- return 0;
+ if (!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
+ time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
+ ((int64_t) c->fmt_in->pts_num*1000);
+ delta_pts = c->cur_pts - time_pts;
+ m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
+ return m_delay>0 ? m_delay : 0;
+ } else {
+ return 0;
+ }
}
#endif
int i, len, ret;
AVFormatContext *ctx;
+ av_freep(&c->pb_buffer);
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
sizeof(AVStream));
st->codec.frame_number = 0; /* XXX: should be done in
AVStream, not in codec */
+ /* I'm pretty sure that this is not correct...
+ * However, without it, we crash
+ */
+ st->codec.coded_frame = &dummy_frame;
}
c->got_key_frame = 0;
}
c->fmt_ctx.pb.is_streamed = 1;
+ av_set_parameters(&c->fmt_ctx, NULL);
av_write_header(&c->fmt_ctx);
len = url_close_dyn_buf(&c->fmt_ctx.pb, &c->pb_buffer);
return 1; /* state changed */
}
}
+ redo:
if (av_read_frame(c->fmt_in, &pkt) < 0) {
if (c->stream->feed && c->stream->feed->feed_opened) {
/* if coming from feed, it means we reached the end of the
c->state = HTTPSTATE_WAIT_FEED;
return 1; /* state changed */
} else {
- /* must send trailer now because eof or error */
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ if (c->stream->loop) {
+ av_close_input_file(c->fmt_in);
+ c->fmt_in = NULL;
+ if (open_input_stream(c, "") < 0)
+ goto no_loop;
+ goto redo;
+ } else {
+ no_loop:
+ /* must send trailer now because eof or error */
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
}
} else {
/* update first pts if needed */
- if (c->first_pts == AV_NOPTS_VALUE)
+ if (c->first_pts == AV_NOPTS_VALUE) {
c->first_pts = pkt.pts;
-
+ c->start_time = cur_time;
+ }
+ c->cur_pts = pkt.pts;
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
if (c->is_packetized) {
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
+ if(!ctx) {
+ av_free_packet(&pkt);
+ break;
+ }
codec = &ctx->streams[0]->codec;
+ /* only one stream per RTP connection */
+ pkt.stream_index = 0;
} else {
ctx = &c->fmt_ctx;
/* Fudge here */
codec = &ctx->streams[pkt.stream_index]->codec;
}
- codec->key_frame = ((pkt.flags & PKT_FLAG_KEY) != 0);
+ codec->coded_frame->key_frame = ((pkt.flags & PKT_FLAG_KEY) != 0);
#ifdef PJSG
if (codec->codec_type == CODEC_TYPE_AUDIO) {
#endif
if (c->is_packetized) {
- ret = url_open_dyn_packet_buf(&ctx->pb,
- url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]));
+ int max_packet_size;
+ if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ else
+ max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
+ ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
c->packet_byte_count = 0;
c->packet_start_time_us = av_gettime();
} else {
#define SHORT_TERM_BANDWIDTH 8000000
/* should convert the format at the same time */
+/* send data starting at c->buffer_ptr to the output connection
+ (either UDP or TCP connection) */
static int http_send_data(HTTPContext *c)
{
int len, ret, dt;
-
- while (c->buffer_ptr >= c->buffer_end) {
- av_freep(&c->pb_buffer);
- ret = http_prepare_data(c);
- if (ret < 0)
- return -1;
- else if (ret == 0) {
- continue;
- } else {
- /* state change requested */
- return 0;
- }
- }
- if (c->buffer_ptr < c->buffer_end) {
- if (c->is_packetized) {
- /* RTP/UDP data output */
- len = c->buffer_end - c->buffer_ptr;
- if (len < 4) {
- /* fail safe - should never happen */
- fail1:
- c->buffer_ptr = c->buffer_end;
- return 0;
- }
- len = (c->buffer_ptr[0] << 24) |
- (c->buffer_ptr[1] << 16) |
- (c->buffer_ptr[2] << 8) |
- (c->buffer_ptr[3]);
- if (len > (c->buffer_end - c->buffer_ptr))
- goto fail1;
-
- /* short term bandwidth limitation */
- dt = av_gettime() - c->packet_start_time_us;
- if (dt < 1)
- dt = 1;
-
- if ((c->packet_byte_count + len) * (INT64)1000000 >=
- (SHORT_TERM_BANDWIDTH / 8) * (INT64)dt) {
- /* bandwidth overflow : wait at most one tick and retry */
- c->state = HTTPSTATE_WAIT_SHORT;
- return 0;
+ for(;;) {
+ if (c->buffer_ptr >= c->buffer_end) {
+ ret = http_prepare_data(c);
+ if (ret < 0)
+ return -1;
+ else if (ret != 0) {
+ /* state change requested */
+ break;
}
-
- c->buffer_ptr += 4;
- url_write(c->rtp_handles[c->packet_stream_index],
- c->buffer_ptr, len);
- c->buffer_ptr += len;
- c->packet_byte_count += len;
} else {
- /* TCP data output */
- len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
- if (len < 0) {
- if (errno != EAGAIN && errno != EINTR) {
- /* error : close connection */
- return -1;
- } else {
+ if (c->is_packetized) {
+ /* RTP data output */
+ len = c->buffer_end - c->buffer_ptr;
+ if (len < 4) {
+ /* fail safe - should never happen */
+ fail1:
+ c->buffer_ptr = c->buffer_end;
return 0;
}
- } else {
+ len = (c->buffer_ptr[0] << 24) |
+ (c->buffer_ptr[1] << 16) |
+ (c->buffer_ptr[2] << 8) |
+ (c->buffer_ptr[3]);
+ if (len > (c->buffer_end - c->buffer_ptr))
+ goto fail1;
+
+ if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
+ /* RTP packets are sent inside the RTSP TCP connection */
+ ByteIOContext pb1, *pb = &pb1;
+ int interleaved_index, size;
+ uint8_t header[4];
+ HTTPContext *rtsp_c;
+
+ rtsp_c = c->rtsp_c;
+ /* if no RTSP connection left, error */
+ if (!rtsp_c)
+ return -1;
+ /* if already sending something, then wait. */
+ if (rtsp_c->state != RTSPSTATE_WAIT_REQUEST) {
+ break;
+ }
+ if (url_open_dyn_buf(pb) < 0)
+ goto fail1;
+ interleaved_index = c->packet_stream_index * 2;
+ /* RTCP packets are sent at odd indexes */
+ if (c->buffer_ptr[1] == 200)
+ interleaved_index++;
+ /* write RTSP TCP header */
+ header[0] = '$';
+ header[1] = interleaved_index;
+ header[2] = len >> 8;
+ header[3] = len;
+ put_buffer(pb, header, 4);
+ /* write RTP packet data */
+ c->buffer_ptr += 4;
+ put_buffer(pb, c->buffer_ptr, len);
+ size = url_close_dyn_buf(pb, &c->packet_buffer);
+ /* prepare asynchronous TCP sending */
+ rtsp_c->packet_buffer_ptr = c->packet_buffer;
+ rtsp_c->packet_buffer_end = c->packet_buffer + size;
+ rtsp_c->state = RTSPSTATE_SEND_PACKET;
+ } else {
+ /* send RTP packet directly in UDP */
+
+ /* short term bandwidth limitation */
+ dt = av_gettime() - c->packet_start_time_us;
+ if (dt < 1)
+ dt = 1;
+
+ if ((c->packet_byte_count + len) * (int64_t)1000000 >=
+ (SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
+ /* bandwidth overflow : wait at most one tick and retry */
+ c->state = HTTPSTATE_WAIT_SHORT;
+ return 0;
+ }
+
+ c->buffer_ptr += 4;
+ url_write(c->rtp_handles[c->packet_stream_index],
+ c->buffer_ptr, len);
+ }
c->buffer_ptr += len;
+ c->packet_byte_count += len;
+ } else {
+ /* TCP data output */
+ len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ return -1;
+ } else {
+ return 0;
+ }
+ } else {
+ c->buffer_ptr += len;
+ }
}
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+ break;
}
- c->data_count += len;
- update_datarate(&c->datarate, c->data_count);
- if (c->stream)
- c->stream->bytes_served += len;
- }
+ } /* for(;;) */
return 0;
}
if (c->stream->feed_opened)
return -1;
+ /* Don't permit writing to this one */
+ if (c->stream->readonly)
+ return -1;
+
/* open feed */
fd = open(c->stream->feed_filename, O_RDWR);
if (fd < 0)
if (!fmt_in)
goto fail;
- s.priv_data = av_mallocz(fmt_in->priv_data_size);
- if (!s.priv_data)
- goto fail;
+ if (fmt_in->priv_data_size > 0) {
+ s.priv_data = av_mallocz(fmt_in->priv_data_size);
+ if (!s.priv_data)
+ goto fail;
+ } else
+ s.priv_data = NULL;
if (fmt_in->read_header(&s, 0) < 0) {
av_freep(&s.priv_data);
if (!strcmp(cmd, "DESCRIBE")) {
rtsp_cmd_describe(c, url);
+ } else if (!strcmp(cmd, "OPTIONS")) {
+ rtsp_cmd_options(c, url);
} else if (!strcmp(cmd, "SETUP")) {
rtsp_cmd_setup(c, url, header);
} else if (!strcmp(cmd, "PLAY")) {
return 0;
}
-static int prepare_sdp_description(HTTPContext *c,
- FFStream *stream, UINT8 **pbuffer)
+/* XXX: move that to rtsp.c, but would need to replace FFStream by
+ AVFormatContext */
+static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
+ struct in_addr my_ip)
{
ByteIOContext pb1, *pb = &pb1;
- struct sockaddr_in my_addr;
- int len, i, payload_type;
+ int i, payload_type, port, private_payload_type, j;
const char *ipstr, *title, *mediatype;
AVStream *st;
- len = sizeof(my_addr);
- getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
- ipstr = inet_ntoa(my_addr.sin_addr);
-
if (url_open_dyn_buf(pb) < 0)
return -1;
/* general media info */
url_fprintf(pb, "v=0\n");
+ ipstr = inet_ntoa(my_ip);
url_fprintf(pb, "o=- 0 0 IN IP4 %s\n", ipstr);
title = stream->title;
if (title[0] == '\0')
url_fprintf(pb, "s=%s\n", title);
if (stream->comment[0] != '\0')
url_fprintf(pb, "i=%s\n", stream->comment);
-
+ if (stream->is_multicast) {
+ url_fprintf(pb, "c=IN IP4 %s\n", inet_ntoa(stream->multicast_ip));
+ }
/* for each stream, we output the necessary info */
+ private_payload_type = 96;
for(i = 0; i < stream->nb_streams; i++) {
st = stream->streams[i];
switch(st->codec.codec_type) {
mediatype = "application";
break;
}
- /* XXX: the port indication is not correct (but should be correct
- for broadcast) */
+ /* NOTE: the port indication is not correct in case of
+ unicast. It is not an issue because RTSP gives it */
payload_type = rtp_get_payload_type(&st->codec);
-
+ if (payload_type < 0)
+ payload_type = private_payload_type++;
+ if (stream->is_multicast) {
+ port = stream->multicast_port + 2 * i;
+ } else {
+ port = 0;
+ }
url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
- mediatype, 0, payload_type);
+ mediatype, port, payload_type);
+ if (payload_type >= 96) {
+ /* for private payload type, we need to give more info */
+ switch(st->codec.codec_id) {
+ case CODEC_ID_MPEG4:
+ {
+ uint8_t *data;
+ url_fprintf(pb, "a=rtpmap:%d MP4V-ES/%d\n",
+ payload_type, 90000);
+ /* we must also add the mpeg4 header */
+ data = st->codec.extradata;
+ if (data) {
+ url_fprintf(pb, "a=fmtp:%d config=", payload_type);
+ for(j=0;j<st->codec.extradata_size;j++) {
+ url_fprintf(pb, "%02x", data[j]);
+ }
+ url_fprintf(pb, "\n");
+ }
+ }
+ break;
+ default:
+ /* XXX: add other codecs ? */
+ goto fail;
+ }
+ }
url_fprintf(pb, "a=control:streamid=%d\n", i);
}
return url_close_dyn_buf(pb, pbuffer);
+ fail:
+ url_close_dyn_buf(pb, pbuffer);
+ av_free(*pbuffer);
+ return -1;
+}
+
+static void rtsp_cmd_options(HTTPContext *c, const char *url)
+{
+// rtsp_reply_header(c, RTSP_STATUS_OK);
+ url_fprintf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
+ url_fprintf(c->pb, "CSeq: %d\r\n", c->seq);
+ url_fprintf(c->pb, "Public: %s\r\n", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE");
+ url_fprintf(c->pb, "\r\n");
}
static void rtsp_cmd_describe(HTTPContext *c, const char *url)
FFStream *stream;
char path1[1024];
const char *path;
- UINT8 *content;
- int content_length;
+ uint8_t *content;
+ int content_length, len;
+ struct sockaddr_in my_addr;
/* find which url is asked */
url_split(NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
found:
/* prepare the media description in sdp format */
- content_length = prepare_sdp_description(c, stream, &content);
+
+ /* get the host IP */
+ len = sizeof(my_addr);
+ getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
+
+ content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
if (content_length < 0) {
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
return NULL;
}
-RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol)
+static RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol)
{
RTSPTransportField *th;
int i;
/* find rtp session, and create it if none found */
rtp_c = find_rtp_session(h->session_id);
if (!rtp_c) {
- rtp_c = rtp_new_connection(c, stream, h->session_id);
+ /* always prefer UDP */
+ th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
+ if (!th) {
+ th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
+ if (!th) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+ }
+
+ rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id,
+ th->protocol);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
return;
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
}
-
- /* always prefer UDP */
- th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
- if (!th) {
- th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
- if (!th) {
- rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
- return;
- }
- }
- rtp_c->rtp_protocol = th->protocol;
}
/* test if stream is OK (test needed because several SETUP needs
}
/* setup stream */
- if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
HTTPContext *rtp_c;
char path1[1024];
const char *path;
+ char buf[1024];
+ int s;
rtp_c = find_rtp_session(session_id);
if (!rtp_c)
path = path1;
if (*path == '/')
path++;
- if (strcmp(path, rtp_c->stream->filename) != 0)
- return NULL;
- return rtp_c;
+ if(!strcmp(path, rtp_c->stream->filename)) return rtp_c;
+ for(s=0; s<rtp_c->stream->nb_streams; ++s) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ rtp_c->stream->filename, s);
+ if(!strncmp(path, buf, sizeof(buf))) {
+ // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
+ return rtp_c;
+ }
+ }
+ return NULL;
}
static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h)
}
rtp_c->state = HTTPSTATE_READY;
-
+ rtp_c->first_pts = AV_NOPTS_VALUE;
/* now everything is OK, so we can send the connection parameters */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
/********************************************************************/
/* RTP handling */
-static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
- FFStream *stream, const char *session_id)
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol)
{
HTTPContext *c = NULL;
-
+ const char *proto_str;
+
/* XXX: should output a warning page when coming
close to the connection limit */
if (nb_connections >= nb_max_connections)
c->fd = -1;
c->poll_entry = NULL;
- c->from_addr = rtsp_c->from_addr;
+ c->from_addr = *from_addr;
c->buffer_size = IOBUFFER_INIT_SIZE;
c->buffer = av_malloc(c->buffer_size);
if (!c->buffer)
pstrcpy(c->session_id, sizeof(c->session_id), session_id);
c->state = HTTPSTATE_READY;
c->is_packetized = 1;
+ c->rtp_protocol = rtp_protocol;
+
/* protocol is shown in statistics */
- pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ proto_str = "MCAST";
+ break;
+ case RTSP_PROTOCOL_RTP_UDP:
+ proto_str = "UDP";
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ proto_str = "TCP";
+ break;
+ default:
+ proto_str = "???";
+ break;
+ }
+ pstrcpy(c->protocol, sizeof(c->protocol), "RTP/");
+ pstrcat(c->protocol, sizeof(c->protocol), proto_str);
+
+ current_bandwidth += stream->bandwidth;
c->next = first_http_ctx;
first_http_ctx = c;
}
/* add a new RTP stream in an RTP connection (used in RTSP SETUP
- command). if dest_addr is NULL, then TCP tunneling in RTSP is
+ command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
used. */
static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr)
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c)
{
AVFormatContext *ctx;
AVStream *st;
char *ipaddr;
URLContext *h;
- UINT8 *dummy_buf;
-
+ uint8_t *dummy_buf;
+ char buf2[32];
+ int max_packet_size;
+
/* now we can open the relevant output stream */
ctx = av_mallocz(sizeof(AVFormatContext));
if (!ctx)
sizeof(AVStream));
}
- if (dest_addr) {
- /* build destination RTP address */
- ipaddr = inet_ntoa(dest_addr->sin_addr);
-
- snprintf(ctx->filename, sizeof(ctx->filename),
- "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
+ /* build destination RTP address */
+ ipaddr = inet_ntoa(dest_addr->sin_addr);
+
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP:
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ /* RTP/UDP case */
- printf("open %s\n", ctx->filename);
+ /* XXX: also pass as parameter to function ? */
+ if (c->stream->is_multicast) {
+ int ttl;
+ ttl = c->stream->multicast_ttl;
+ if (!ttl)
+ ttl = 16;
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d?multicast=1&ttl=%d",
+ ipaddr, ntohs(dest_addr->sin_port), ttl);
+ } else {
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
+ }
if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
goto fail;
c->rtp_handles[stream_index] = h;
- } else {
+ max_packet_size = url_get_max_packet_size(h);
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ /* RTP/TCP case */
+ c->rtsp_c = rtsp_c;
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ break;
+ default:
goto fail;
}
+ http_log("%s:%d - - [%s] \"PLAY %s/streamid=%d %s\"\n",
+ ipaddr, ntohs(dest_addr->sin_port),
+ ctime1(buf2),
+ c->stream->filename, stream_index, c->protocol);
+
/* normally, no packets should be output here, but the packet size may be checked */
- if (url_open_dyn_packet_buf(&ctx->pb,
- url_get_max_packet_size(h)) < 0) {
+ if (url_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
/* XXX: close stream */
goto fail;
}
+ av_set_parameters(ctx, NULL);
if (av_write_header(ctx) < 0) {
fail:
if (h)
/********************************************************************/
/* ffserver initialization */
-AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec)
+static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec)
{
AVStream *fst;
return NULL;
fst->priv_data = av_mallocz(sizeof(FeedData));
memcpy(&fst->codec, codec, sizeof(AVCodecContext));
+ fst->codec.coded_frame = &dummy_frame;
stream->streams[stream->nb_streams++] = fst;
return fst;
}
/* return the stream number in the feed */
-int add_av_stream(FFStream *feed,
- AVStream *st)
+static int add_av_stream(FFStream *feed, AVStream *st)
{
AVStream *fst;
AVCodecContext *av, *av1;
if (av1->width == av->width &&
av1->height == av->height &&
av1->frame_rate == av->frame_rate &&
+ av1->frame_rate_base == av->frame_rate_base &&
av1->gop_size == av->gop_size)
goto found;
break;
return i;
}
-void remove_stream(FFStream *stream)
+static void remove_stream(FFStream *stream)
{
FFStream **ps;
ps = &first_stream;
}
}
+/* specific mpeg4 handling : we extract the raw parameters */
+static void extract_mpeg4_header(AVFormatContext *infile)
+{
+ int mpeg4_count, i, size;
+ AVPacket pkt;
+ AVStream *st;
+ const uint8_t *p;
+
+ mpeg4_count = 0;
+ for(i=0;i<infile->nb_streams;i++) {
+ st = infile->streams[i];
+ if (st->codec.codec_id == CODEC_ID_MPEG4 &&
+ st->codec.extradata_size == 0) {
+ mpeg4_count++;
+ }
+ }
+ if (!mpeg4_count)
+ return;
+
+ printf("MPEG4 without extra data: trying to find header\n");
+ while (mpeg4_count > 0) {
+ if (av_read_packet(infile, &pkt) < 0)
+ break;
+ st = infile->streams[pkt.stream_index];
+ if (st->codec.codec_id == CODEC_ID_MPEG4 &&
+ st->codec.extradata_size == 0) {
+ av_freep(&st->codec.extradata);
+ /* fill extradata with the header */
+ /* XXX: we make hard suppositions here ! */
+ p = pkt.data;
+ while (p < pkt.data + pkt.size - 4) {
+ /* stop when vop header is found */
+ if (p[0] == 0x00 && p[1] == 0x00 &&
+ p[2] == 0x01 && p[3] == 0xb6) {
+ size = p - pkt.data;
+ // av_hex_dump(pkt.data, size);
+ st->codec.extradata = av_malloc(size);
+ st->codec.extradata_size = size;
+ memcpy(st->codec.extradata, pkt.data, size);
+ break;
+ }
+ p++;
+ }
+ mpeg4_count--;
+ }
+ av_free_packet(&pkt);
+ }
+}
+
/* compute the needed AVStream for each file */
-void build_file_streams(void)
+static void build_file_streams(void)
{
FFStream *stream, *stream_next;
AVFormatContext *infile;
av_close_input_file(infile);
goto fail;
}
+ extract_mpeg4_header(infile);
+
for(i=0;i<infile->nb_streams;i++) {
add_av_stream1(stream, &infile->streams[i]->codec);
}
}
/* compute the needed AVStream for each feed */
-void build_feed_streams(void)
+static void build_feed_streams(void)
{
FFStream *stream, *feed;
int i;
matches = 0;
} else if (ccf->codec_type == CODEC_TYPE_VIDEO) {
if (CHECK_CODEC(frame_rate) ||
+ CHECK_CODEC(frame_rate_base) ||
CHECK_CODEC(width) ||
CHECK_CODEC(height)) {
printf("Codec width, height and framerate do not match for stream %d\n", i);
printf("Deleting feed file '%s' as it appears to be corrupt\n",
feed->feed_filename);
}
- if (!matches)
+ if (!matches) {
+ if (feed->readonly) {
+ printf("Unable to delete feed file '%s' as it is marked readonly\n",
+ feed->feed_filename);
+ exit(1);
+ }
unlink(feed->feed_filename);
+ }
}
if (!url_exist(feed->feed_filename)) {
AVFormatContext s1, *s = &s1;
+ if (feed->readonly) {
+ printf("Unable to create feed file '%s' as it is marked readonly\n",
+ feed->feed_filename);
+ exit(1);
+ }
+
/* only write the header of the ffm file */
if (url_fopen(&s->pb, feed->feed_filename, URL_WRONLY) < 0) {
fprintf(stderr, "Could not open output feed file '%s'\n",
st = feed->streams[i];
s->streams[i] = st;
}
+ av_set_parameters(s, NULL);
av_write_header(s);
/* XXX: need better api */
av_freep(&s->priv_data);
}
}
+/* compute the bandwidth used by each stream */
+static void compute_bandwidth(void)
+{
+ int bandwidth, i;
+ FFStream *stream;
+
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ bandwidth = 0;
+ for(i=0;i<stream->nb_streams;i++) {
+ AVStream *st = stream->streams[i];
+ switch(st->codec.codec_type) {
+ case CODEC_TYPE_AUDIO:
+ case CODEC_TYPE_VIDEO:
+ bandwidth += st->codec.bit_rate;
+ break;
+ default:
+ break;
+ }
+ }
+ stream->bandwidth = (bandwidth + 999) / 1000;
+ }
+}
+
static void get_arg(char *buf, int buf_size, const char **pp)
{
const char *p;
}
/* add a codec and set the default parameters */
-void add_codec(FFStream *stream, AVCodecContext *av)
+static void add_codec(FFStream *stream, AVCodecContext *av)
{
AVStream *st;
case CODEC_TYPE_VIDEO:
if (av->bit_rate == 0)
av->bit_rate = 64000;
- if (av->frame_rate == 0)
- av->frame_rate = 5 * FRAME_RATE_BASE;
+ if (av->frame_rate == 0){
+ av->frame_rate = 5;
+ av->frame_rate_base = 1;
+ }
if (av->width == 0 || av->height == 0) {
av->width = 160;
av->height = 128;
memcpy(&st->codec, av, sizeof(AVCodecContext));
}
-int opt_audio_codec(const char *arg)
+static int opt_audio_codec(const char *arg)
{
AVCodec *p;
return p->id;
}
-int opt_video_codec(const char *arg)
+static int opt_video_codec(const char *arg)
{
AVCodec *p;
/* simplistic plugin support */
+#ifdef CONFIG_HAVE_DLOPEN
void load_module(const char *filename)
{
void *dll;
init_func();
}
+#endif
-int parse_ffconfig(const char *filename)
+static int parse_ffconfig(const char *filename)
{
FILE *f;
char line[1024];
filename, line_num, arg);
errors++;
} else {
- nb_max_bandwidth = val;
+ max_bandwidth = val;
}
} else if (!strcasecmp(cmd, "CustomLog")) {
get_arg(logfilename, sizeof(logfilename), &p);
if (!argbuf[0])
break;
- feed->child_argv[i] = av_malloc(strlen(argbuf + 1));
+ feed->child_argv[i] = av_malloc(strlen(argbuf) + 1);
strcpy(feed->child_argv[i], argbuf);
}
snprintf(feed->child_argv[i], 256, "http://127.0.0.1:%d/%s",
ntohs(my_http_addr.sin_port), feed->filename);
}
+ } else if (!strcasecmp(cmd, "ReadOnlyFile")) {
+ if (feed) {
+ get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
+ feed->readonly = 1;
+ } else if (stream) {
+ get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
+ }
} else if (!strcasecmp(cmd, "File")) {
if (feed) {
get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
fsize *= 1024 * 1024 * 1024;
break;
}
- feed->feed_max_size = (INT64)fsize;
+ feed->feed_max_size = (int64_t)fsize;
}
} else if (!strcasecmp(cmd, "</Feed>")) {
if (!feed) {
} else if (!strcasecmp(cmd, "AudioQuality")) {
get_arg(arg, sizeof(arg), &p);
if (stream) {
- audio_enc.quality = atof(arg) * 1000;
+// audio_enc.quality = atof(arg) * 1000;
}
} else if (!strcasecmp(cmd, "VideoBitRateRange")) {
if (stream) {
} else if (!strcasecmp(cmd, "VideoFrameRate")) {
get_arg(arg, sizeof(arg), &p);
if (stream) {
- video_enc.frame_rate = (int)(strtod(arg, NULL) * FRAME_RATE_BASE);
+ video_enc.frame_rate_base= DEFAULT_FRAME_RATE_BASE;
+ video_enc.frame_rate = (int)(strtod(arg, NULL) * video_enc.frame_rate_base);
}
} else if (!strcasecmp(cmd, "VideoGopSize")) {
get_arg(arg, sizeof(arg), &p);
}
} else if (!strcasecmp(cmd, "VideoHighQuality")) {
if (stream) {
- video_enc.flags |= CODEC_FLAG_HQ;
+ video_enc.mb_decision = FF_MB_DECISION_BITS;
+ }
+ } else if (!strcasecmp(cmd, "Video4MotionVector")) {
+ if (stream) {
+ video_enc.mb_decision = FF_MB_DECISION_BITS; //FIXME remove
+ video_enc.flags |= CODEC_FLAG_4MV;
}
} else if (!strcasecmp(cmd, "VideoQDiff")) {
get_arg(arg, sizeof(arg), &p);
errors++;
} else {
/* Only take the first */
- acl.first = *(struct in_addr *) he->h_addr_list[0];
+ acl.first.s_addr = ntohl(((struct in_addr *) he->h_addr_list[0])->s_addr);
acl.last = acl.first;
}
errors++;
} else {
/* Only take the first */
- acl.last = *(struct in_addr *) he->h_addr_list[0];
+ acl.last.s_addr = ntohl(((struct in_addr *) he->h_addr_list[0])->s_addr);
}
}
strcpy(stream->rtsp_option, arg);
}
}
+ } else if (!strcasecmp(cmd, "MulticastAddress")) {
+ get_arg(arg, sizeof(arg), &p);
+ if (stream) {
+ if (!inet_aton(arg, &stream->multicast_ip)) {
+ fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
+ filename, line_num, arg);
+ errors++;
+ }
+ stream->is_multicast = 1;
+ stream->loop = 1; /* default is looping */
+ }
+ } else if (!strcasecmp(cmd, "MulticastPort")) {
+ get_arg(arg, sizeof(arg), &p);
+ if (stream) {
+ stream->multicast_port = atoi(arg);
+ }
+ } else if (!strcasecmp(cmd, "MulticastTTL")) {
+ get_arg(arg, sizeof(arg), &p);
+ if (stream) {
+ stream->multicast_ttl = atoi(arg);
+ }
+ } else if (!strcasecmp(cmd, "NoLoop")) {
+ if (stream) {
+ stream->loop = 0;
+ }
} else if (!strcasecmp(cmd, "</Stream>")) {
if (!stream) {
fprintf(stderr, "%s:%d: No corresponding <Stream> for </Stream>\n",
redirect = NULL;
} else if (!strcasecmp(cmd, "LoadModule")) {
get_arg(arg, sizeof(arg), &p);
+#ifdef CONFIG_HAVE_DLOPEN
load_module(arg);
+#else
+ fprintf(stderr, "%s:%d: Module support not compiled into this version: '%s'\n",
+ filename, line_num, arg);
+ errors++;
+#endif
} else {
fprintf(stderr, "%s:%d: Incorrect keyword: '%s'\n",
filename, line_num, cmd);
#if 0
static void write_packet(FFCodec *ffenc,
- UINT8 *buf, int size)
+ uint8_t *buf, int size)
{
PacketHeader hdr;
AVCodecContext *enc = &ffenc->enc;
- UINT8 *wptr;
+ uint8_t *wptr;
mk_header(&hdr, enc, size);
wptr = http_fifo.wptr;
- fifo_write(&http_fifo, (UINT8 *)&hdr, sizeof(hdr), &wptr);
+ fifo_write(&http_fifo, (uint8_t *)&hdr, sizeof(hdr), &wptr);
fifo_write(&http_fifo, buf, size, &wptr);
/* atomic modification of wptr */
http_fifo.wptr = wptr;
}
#endif
-void help(void)
+static void show_banner(void)
{
- printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000, 2001, 2002 Fabrice Bellard\n"
- "usage: ffserver [-L] [-h] [-f configfile]\n"
+ printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000-2003 Fabrice Bellard\n");
+}
+
+static void show_help(void)
+{
+ show_banner();
+ printf("usage: ffserver [-L] [-h] [-f configfile]\n"
"Hyper fast multi format Audio/Video streaming server\n"
"\n"
- "-L : print the LICENCE\n"
+ "-L : print the LICENSE\n"
"-h : this help\n"
"-f configfile : use configfile instead of /etc/ffserver.conf\n"
);
}
-void licence(void)
+static void show_license(void)
{
+ show_banner();
printf(
- "ffserver version " FFMPEG_VERSION "\n"
- "Copyright (c) 2000, 2001, 2002 Fabrice Bellard\n"
"This library is free software; you can redistribute it and/or\n"
"modify it under the terms of the GNU Lesser General Public\n"
"License as published by the Free Software Foundation; either\n"
break;
switch(c) {
case 'L':
- licence();
+ show_license();
exit(1);
case '?':
case 'h':
- help();
+ show_help();
exit(1);
case 'n':
no_launch = 1;
my_rtsp_addr.sin_addr.s_addr = htonl (INADDR_ANY);
nb_max_connections = 5;
- nb_max_bandwidth = 1000;
+ max_bandwidth = 1000;
first_stream = NULL;
logfilename[0] = '\0';
build_feed_streams();
+ compute_bandwidth();
+
/* put the process in background and detach it from its TTY */
if (ffserver_daemon) {
int pid;
projects / ffmpeg / blobdiff